| 6955870806624479723addfae6dcf5d13968796c |
|
13-Jan-2016 |
Peter Kasting <pkasting@google.com> |
Convert channel counts to size_t.
IIRC, this was originally requested by ajm during review of the other size_t conversions I did over the past year, and I agreed it made sense, but wanted to do it separately since those changes were already gargantuan.
BUG=chromium:81439
TEST=none
R=henrik.lundin@webrtc.org, henrika@webrtc.org, kjellander@webrtc.org, minyue@webrtc.org, perkj@webrtc.org, solenberg@webrtc.org, stefan@webrtc.org, tina.legrand@webrtc.org
Review URL: https://codereview.webrtc.org/1316523002 .
Cr-Commit-Position: refs/heads/master@{#11229}
/external/webrtc/webrtc/modules/audio_processing/audio_buffer.cc
|
| b2328d11dcc86fba1661ee3fa0d51fc126939764 |
|
12-Jan-2016 |
aluebs <aluebs@webrtc.org> |
Remove additional channel constraints when Beamforming is enabled in AudioProcessing
The general constraints on number of channels for AudioProcessing is:
num_in_channels == num_out_channels || num_out_channels == 1
When Beamforming is enabled and additional constraint was added forcing:
num_out_channels == 1
This artificial constraint was removed by adding upmixing support in CopyTo, since it was already supported for the AudioFrame interface using InterleaveTo.
Review URL: https://codereview.webrtc.org/1571013002
Cr-Commit-Position: refs/heads/master@{#11215}
/external/webrtc/webrtc/modules/audio_processing/audio_buffer.cc
|
| 25702cb1628941427fa55e528f53483f239ae011 |
|
08-Jan-2016 |
pkasting <pkasting@chromium.org> |
Misc. small cleanups.
* Better param names
* Avoid using negative values for (bogus) placeholder channel counts (mostly in tests). Since channels will be changing to size_t, negative values will be illegal; it's sufficient to use 0 in these cases.
* Use arraysize()
* Use size_t for counting frames, samples, blocks, buffers, and bytes -- most of these are already size_t in most places, this just fixes some stragglers
* reinterpret_cast<int64_t>(void*) is not necessarily safe; use uintptr_t instead
* Remove unnecessary code, e.g. dead code, needlessly long/repetitive code, or function overrides that exactly match the base definition
* Fix indenting
* Use uint32_t for timestamps (matching how it's already a uint32_t in most places)
* Spelling
* RTC_CHECK_EQ(expected, actual)
* Rewrap
* Use .empty()
* Be more pedantic about matching int/int32_t/
* Remove pointless consts on input parameters to functions
* Add missing sanity checks
All this was found in the course of constructing https://codereview.webrtc.org/1316523002/ , and is being landed separately first.
BUG=none
TEST=none
Review URL: https://codereview.webrtc.org/1534193008
Cr-Commit-Position: refs/heads/master@{#11191}
/external/webrtc/webrtc/modules/audio_processing/audio_buffer.cc
|
| dce40cf804019a9898b6ab8d8262466b697c56e0 |
|
24-Aug-2015 |
Peter Kasting <pkasting@google.com> |
Update a ton of audio code to use size_t more correctly and in general reduce
use of int16_t/uint16_t.
This is the upshot of a recommendation by henrik.lundin and kwiberg on an original small change ( https://webrtc-codereview.appspot.com/42569004/#ps1 ) to stop using int16_t just because values could fit in it, and is similar in nature to a previous "mass change to use size_t more" ( https://webrtc-codereview.appspot.com/23129004/ ) which also needed to be split up for review but to land all at once, since, like adding "const", such changes tend to cause a lot of transitive effects.
This was be reviewed and approved in pieces:
https://codereview.webrtc.org/1224093003
https://codereview.webrtc.org/1224123002
https://codereview.webrtc.org/1224163002
https://codereview.webrtc.org/1225133003
https://codereview.webrtc.org/1225173002
https://codereview.webrtc.org/1227163003
https://codereview.webrtc.org/1227203003
https://codereview.webrtc.org/1227213002
https://codereview.webrtc.org/1227893002
https://codereview.webrtc.org/1228793004
https://codereview.webrtc.org/1228803003
https://codereview.webrtc.org/1228823002
https://codereview.webrtc.org/1228823003
https://codereview.webrtc.org/1228843002
https://codereview.webrtc.org/1230693002
https://codereview.webrtc.org/1231713002
The change is being landed as TBR to all the folks who reviewed the above.
BUG=chromium:81439
TEST=none
R=andrew@webrtc.org, pbos@webrtc.org
TBR=aluebs, andrew, asapersson, henrika, hlundin, jan.skoglund, kwiberg, minyue, pbos, pthatcher
Review URL: https://codereview.webrtc.org/1230503003 .
Cr-Commit-Position: refs/heads/master@{#9768}
/external/webrtc/webrtc/modules/audio_processing/audio_buffer.cc
|
| 60d9b332a5391045439bfb6a3a5447973e3d5603 |
|
14-Aug-2015 |
ekmeyerson <ekmeyerson@webrtc.org> |
Integrate Intelligibility with APM
- Integrates intelligibility into audio_processing.
- Allows modification of reverse stream if intelligibility enabled.
- Makes intelligibility available in audioproc_float test.
- Adds reverse stream processing to audioproc_float.
- (removed) Makes intelligibility toggleable in real time in voe_cmd_test.
- Cleans up intelligibility construction, parameters, constants and dead code.
TBR=pbos@webrtc.org
Review URL: https://codereview.webrtc.org/1234463003
Cr-Commit-Position: refs/heads/master@{#9713}
/external/webrtc/webrtc/modules/audio_processing/audio_buffer.cc
|
| 86c6d33aec684d08189d498912e47cbc17c4d2db |
|
23-Jul-2015 |
Michael Graczyk <mgraczyk@chromium.org> |
Allow more than 2 input channels in AudioProcessing.
The number of output channels is constrained to be equal to either 1 or the
number of input channels.
An earlier version of this commit caused a crash on AEC dump.
TBR=aluebs@webrtc.org,pbos@webrtc.org
Review URL: https://codereview.webrtc.org/1248393003 .
Cr-Commit-Position: refs/heads/master@{#9626}
/external/webrtc/webrtc/modules/audio_processing/audio_buffer.cc
|
| 64e753c3998a17429418180b3a947231a9fd98cd |
|
23-Jul-2015 |
magjed <magjed@webrtc.org> |
Revert of Allow more than 2 input channels in AudioProcessing. (patchset #13 id:240001 of https://codereview.webrtc.org/1226093007/)
Reason for revert:
Breaks Chromium FYI content_browsertest on all platforms. The testcase that fails is WebRtcAecDumpBrowserTest.CallWithAecDump.
https://build.chromium.org/p/chromium.webrtc.fyi/builders/Linux/builds/19388
Sample output:
[ RUN ] WebRtcAecDumpBrowserTest.CallWithAecDump
Xlib: extension "RANDR" missing on display ":9".
[4:14:0722/211548:1282124453:WARNING:webrtcvoiceengine.cc(472)] Unexpected codec: ISAC/48000/1 (105)
[4:14:0722/211548:1282124593:WARNING:webrtcvoiceengine.cc(472)] Unexpected codec: PCMU/8000/2 (110)
[4:14:0722/211548:1282124700:WARNING:webrtcvoiceengine.cc(472)] Unexpected codec: PCMA/8000/2 (118)
[4:14:0722/211548:1282124815:WARNING:webrtcvoiceengine.cc(472)] Unexpected codec: G722/8000/2 (119)
[19745:19745:0722/211548:1282133667:INFO:CONSOLE(64)] "Looking at video in element remote-view-1", source: http://127.0.0.1:48819/media/webrtc_test_utilities.js (64)
[19745:19745:0722/211548:1282136892:INFO:CONSOLE(64)] "Looking at video in element remote-view-2", source: http://127.0.0.1:48819/media/webrtc_test_utilities.js (64)
../../content/test/webrtc_content_browsertest_base.cc:62: Failure
Value of: ExecuteScriptAndExtractString( shell()->web_contents(), javascript, &result)
Actual: false
Expected: true
Failed to execute javascript call({video: true, audio: true});.
From javascript: (nothing)
When executing 'call({video: true, audio: true});'
../../content/test/webrtc_content_browsertest_base.cc:75: Failure
Failed
../../content/browser/media/webrtc_aecdump_browsertest.cc:26: Failure
Expected: (base::kNullProcessId) != (*id), actual: 0 vs 0
../../content/browser/media/webrtc_aecdump_browsertest.cc:95: Failure
Value of: GetRenderProcessHostId(&render_process_id)
Actual: false
Expected: true
../../content/browser/media/webrtc_aecdump_browsertest.cc:99: Failure
Value of: base::PathExists(dump_file)
Actual: false
Expected: true
../../content/browser/media/webrtc_aecdump_browsertest.cc:101: Failure
Value of: base::GetFileSize(dump_file, &file_size)
Actual: false
Expected: true
../../content/browser/media/webrtc_aecdump_browsertest.cc:102: Failure
Expected: (file_size) > (0), actual: 0 vs 0
[ FAILED ] WebRtcAecDumpBrowserTest.CallWithAecDump, where TypeParam = and GetParam() = (361 ms)
Original issue's description:
> Allow more than 2 input channels in AudioProcessing.
>
> The number of output channels is constrained to be equal to either 1 or the
> number of input channels.
>
> R=aluebs@webrtc.org, andrew@webrtc.org, pbos@webrtc.org
>
> Committed: https://chromium.googlesource.com/external/webrtc/+/c204754b7a0cc801c70e8ce6c689f57f6ce00b3b
TBR=andrew@webrtc.org,aluebs@webrtc.org,ajm@chromium.org,pbos@chromium.org,pbos@webrtc.org,mgraczyk@chromium.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
Review URL: https://codereview.webrtc.org/1253573005
Cr-Commit-Position: refs/heads/master@{#9621}
/external/webrtc/webrtc/modules/audio_processing/audio_buffer.cc
|
| c204754b7a0cc801c70e8ce6c689f57f6ce00b3b |
|
23-Jul-2015 |
Michael Graczyk <mgraczyk@chromium.org> |
Allow more than 2 input channels in AudioProcessing.
The number of output channels is constrained to be equal to either 1 or the
number of input channels.
R=aluebs@webrtc.org, andrew@webrtc.org, pbos@webrtc.org
Review URL: https://codereview.webrtc.org/1226093007 .
Cr-Commit-Position: refs/heads/master@{#9619}
/external/webrtc/webrtc/modules/audio_processing/audio_buffer.cc
|
| 728d9037c016c01295177fa700fc7927f0bb80bb |
|
11-Jun-2015 |
Peter Kasting <pkasting@google.com> |
Reformat existing code. There should be no functional effects.
This includes changes like:
* Attempt to break lines at better positions
* Use "override" in more places, don't use "virtual" with it
* Use {} where the body is more than one line
* Make declaration and definition arg names match
* Eliminate unused code
* EXPECT_EQ(expected, actual) (but use (actual, expected) for e.g. _GT)
* Correct #include order
* Use anonymous namespaces in preference to "static" for file-scoping
* Eliminate unnecessary casts
* Update reference code in comments of ARM assembly sources to match actual current C code
* Fix indenting to be more style-guide compliant
* Use arraysize() in more places
* Use bool instead of int for "boolean" values (0/1)
* Shorten and simplify code
* Spaces around operators
* 80 column limit
* Use const more consistently
* Space goes after '*' in type name, not before
* Remove unnecessary return values
* Use "(var == const)", not "(const == var)"
* Spelling
* Prefer true, typed constants to "enum hack" constants
* Avoid "virtual" on non-overridden functions
* ASSERT(x == y) -> ASSERT_EQ(y, x)
BUG=none
R=andrew@webrtc.org, asapersson@webrtc.org, henrika@webrtc.org, juberti@webrtc.org, kjellander@webrtc.org, kwiberg@webrtc.org
Review URL: https://codereview.webrtc.org/1172163004
Cr-Commit-Position: refs/heads/master@{#9420}
/external/webrtc/webrtc/modules/audio_processing/audio_buffer.cc
|
| 05c760533f33cdd4f57fc1c1f15030708f6c185b |
|
20-May-2015 |
Alejandro Luebs <aluebs@webrtc.org> |
Add resampling support in AudioBuffer::DeinterleaveFrom
It is necessary for adding 48kHz support to the AudioProcessing::AnalyzeReverseStream int interface (It was not necessary for 32kHz since in that case the splitting filter is more efficient).
BUG=webrtc:3146
R=andrew@webrtc.org, bjornv@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/56389004
Cr-Commit-Position: refs/heads/master@{#9241}
/external/webrtc/webrtc/modules/audio_processing/audio_buffer.cc
|
| 5a92aa844096b056e8c46f28fc7c074b3c3c136d |
|
27-Apr-2015 |
Alejandro Luebs <aluebs@webrtc.org> |
Add 3-band filter-bank implementation
The implementation is a FIR filter bank with DCT modulation, similar to the proposed in "Multirate Signal Processing for Communication Systems" by Fredric J Harris.
The lowpass filter prototype has these characteristics:
* Passband ripple = 0.3dB
* Passband frequency = 0.147 (7kHz at 48kHz)
* Stopband attenuation = 40dB
* Stopband frequency = 0.192 (9.2kHz at 48kHz)
* Delay = 24 samples (500us at 48kHz)
* Linear phase
This filter bank does not satisfy perfect reconstruction. The SNR after analysis and synthesis (with no processing in between) is approximately 9.5dB depending on the input signal after compensating for the delay.
The performance on my workstation of AudioProcessing (with AGC and NS enabled) on a 413s recording compared to previous versions is as follows:
* Input signal has 32kHz sample rate: 3.01s
* Resampling 48kHz to 32kHz: 3.56s
* Today's temporary filter bank: 5.67s
* This filter-bank: 4.62s
BUG=webrtc:3146
R=andrew@webrtc.org, bjornv@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/48999005
Cr-Commit-Position: refs/heads/master@{#9090}
/external/webrtc/webrtc/modules/audio_processing/audio_buffer.cc
|
| 3aca0b0b312a77e9399502a64c27bc3fcb981927 |
|
26-Feb-2015 |
aluebs@webrtc.org <aluebs@webrtc.org> |
Add 48kHz support to Beamformer
Doing something similar for the band 16-24kHz to what is done for the band 8-16kHz
Tested for 32kHz sample rate and the output is bitexact with how it was before this CL.
BUG=webrtc:3146
R=andrew@webrtc.org, kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/35159004
Cr-Commit-Position: refs/heads/master@{#8522}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8522 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/audio_buffer.cc
|
| d35a5c350617cc9d60ce45201764a99229b7299a |
|
10-Feb-2015 |
aluebs@webrtc.org <aluebs@webrtc.org> |
Make ChannelBuffer aware of frequency bands
Now the ChannelBuffer has 2 separate arrays, one for the full-band data and one for the splitted one. The corresponding accessors are added to the ChannelBuffer.
This is done to avoid having to refresh the bands pointers in AudioBuffer. It will also allow us to have a general accessor like data()[band][channel][sample].
All the files using the ChannelBuffer needed to be re-factored.
Tested with modules_unittests, common_audio_unittests, audioproc, audioproc_f, voe_cmd_test.
R=andrew@webrtc.org, kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36999004
Cr-Commit-Position: refs/heads/master@{#8318}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8318 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/audio_buffer.cc
|
| 035e9123e9c7de3b09b37bc8e57907e4af7ce219 |
|
28-Jan-2015 |
kjellander@webrtc.org <kjellander@webrtc.org> |
Move channel_buffer.{h,cc} to common_audio.
In https://code.google.com/p/webrtc/source/detail?r=8166
I added a check preventing GYP files from referencing
sources above their directory level.
This CL fixes the disallowed reference added in
https://code.google.com/p/webrtc/source/detail?r=8157
by moving channel_buffer.{h,cc} to common_audio for real.
BUG=4185
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/35939004
Cr-Commit-Position: refs/heads/master@{#8190}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8190 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/audio_buffer.cc
|
| 27d106bcf7eaf864e8433f1fc303475b953498b3 |
|
11-Dec-2014 |
aluebs@webrtc.org <aluebs@webrtc.org> |
Move the downmixing out of AudioBuffer
This provides more flexibility if some component in AudioProcessing wants to operate before downmixing.
Now the AudioProcessing does only track the processing rate, but not the processing number of channels. This is tracked by the AudioBuffer itself and can be changed at any time to one smaller or equal the input number of channels. For each chunk it is reset to input number of channels and the end it should be equal to the output number of channels.
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/28169004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7879 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/audio_buffer.cc
|
| c5ebbd98f5996db0defbbfc14f5ca41e620bd7e4 |
|
10-Dec-2014 |
aluebs@webrtc.org <aluebs@webrtc.org> |
Support 48kHz in Noise Suppression
Doing the same for the 16-24kHz band than was done in the 8-16kHz.
Results look and sound as nice.
BUG=webrtc:3146
R=andrew@webrtc.org, bjornv@webrtc.org, kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/29139004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7865 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/audio_buffer.cc
|
| a7384a1126cda7ce726f73b023bad997627fc138 |
|
03-Dec-2014 |
aluebs@webrtc.org <aluebs@webrtc.org> |
Simplify audio_buffer APIs
Now there is only one API to get the data or the channels (one const and one no const) merged or by band.
The band is passed in as a parameter, instead of calling different methods.
BUG=webrtc:3146
R=andrew@webrtc.org, bjornv@webrtc.org, kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/27249004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7790 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/audio_buffer.cc
|
| 8789376cd35e055765a72248a8ad444ea2e9438c |
|
28-Nov-2014 |
aluebs@webrtc.org <aluebs@webrtc.org> |
Move ChannelBuffer class to channel_buffer file
No change in functionallity.
BUG=webrtc:3146
R=andrew@webrtc.org, bjornv@webrtc.org, kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/28109004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7760 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/audio_buffer.cc
|
| 79b9eba3abbf02f7a569280d8a735847ce09ec25 |
|
26-Nov-2014 |
aluebs@webrtc.org <aluebs@webrtc.org> |
Implement 3 band splitting filter bank by upsampling and splitting twice into 2 bands
Implemented the 3 bands splitting filter bank by:
1. Upsample by 4/3.
2. Split twice into 2 bands.
3. Discard upper most band, because it is empty anyway.
A unittest was also implemented:
1. Generate a signal from presence or absence of sine waves of different frequencies.
2. Split into 3 bands and check their presence or absence.
3. Recombine the bands.
4. Calculate delay (as it is an IIR it depends on frequency).
5. Check that the cross correlation of input and output is high enough at that delay.
BUG=webrtc:3146
R=andrew@webrtc.org, bjornv@webrtc.org, kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/31029004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7754 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/audio_buffer.cc
|
| 087da13fe8be20b61b556f60eac937c9d0b19fb8 |
|
18-Nov-2014 |
aluebs@webrtc.org <aluebs@webrtc.org> |
Add empty 3 band splitting filter API
This is only an empty API that will never be used. For now is 48kHz not supported in AudioProcessing. For that it needs to be added in InitializeLocked. But before the 3 band filter bank needs to be populated.
BUG=webrtc:3146
R=bjornv@webrtc.org, kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/30139004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7715 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/audio_buffer.cc
|
| be05c74ec8ee975f3451809425756d0f6a51ff2e |
|
14-Nov-2014 |
aluebs@webrtc.org <aluebs@webrtc.org> |
Wrap the splitting filter in its own class
This doesn't change the behavior at all.
The logic behind this is having one class which manages all the splitting filters, because in the future we plan to add a 3 band one for 48kHz support.
It also breaks the dependency of the AudioBuffer with the filter states of these filters (which are going to be different for the 3 band one). The AudioBuffer is complicated enough and is going to need changes to support 3 bands in the future, so any simplification is a good idea.
On top of that it eliminates repeated code in the APM (now only iterating over channels, but then also deciding in how many bands to split). This should be managed by the AudioBuffer directly.
BUG=webrtc:3146
R=bjornv@webrtc.org, kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/32469004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7705 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/audio_buffer.cc
|
| 8328e7c44d59bb9fcbc7f8a033beb3d073929518 |
|
31-Oct-2014 |
andrew@webrtc.org <andrew@webrtc.org> |
Revert "Revert part of r7561, "Refactor audio conversion functions.""
This restores the conversion changes to AudioProcessing originally
added in r7561, with minor alterations to ensure it passes all tests.
TBR=kwiberg
Review URL: https://webrtc-codereview.appspot.com/28899004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7574 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/audio_buffer.cc
|
| bcfb4d0403d9e45e37a4d93de919c2a3df57ce50 |
|
30-Oct-2014 |
kwiberg@webrtc.org <kwiberg@webrtc.org> |
Revert part of r7561, "Refactor audio conversion functions."
Specifically, revert this part:
"Remove hacks in AudioBuffer intended to maintain bit-exactness with
the float path. The conversions etc. are now all natural, and
instead we enforce close but not bit-exact output between the two
paths."
But keep the conversion function rename, since that doesn't seem to be
causing problems.
R=tina.legrand@webrtc.org, bjornv@webrtc.org
TBR=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/24999004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7569 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/audio_buffer.cc
|
| 4fc4addc815379c6cc1d04f4853a528390fe09ba |
|
30-Oct-2014 |
andrew@webrtc.org <andrew@webrtc.org> |
Refactor audio conversion functions.
Use a consistent naming scheme that can be understood at the callsite
without having to refer to documentation.
Remove hacks in AudioBuffer intended to maintain bit-exactness with the
float path. The conversions etc. are now all natural, and instead we
enforce close but not bit-exact output between the two paths.
Output of ApmTest.Process:
https://paste.googleplex.com/5931055831842816
R=aluebs@webrtc.org, bjornv@webrtc.org, kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13049004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7561 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/audio_buffer.cc
|
| bfacaabfce3cee012672d8fcf15326e248261996 |
|
25-Sep-2014 |
claguna@google.com <claguna@google.com> |
Add accessors for array of channel pointers in AudioBuffer. They are
needed as arguments to any multichannel audio processing unit.
R=andrew@webrtc.org, kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/30499004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7303 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/audio_buffer.cc
|
| 30be827e6a2bfae76c445e62d0853f83d238814a |
|
24-Sep-2014 |
andrew@webrtc.org <andrew@webrtc.org> |
Enable render downmixing to mono in AudioProcessing.
In practice, we have been doing this since time immemorial, but have
relied on the user to do the downmixing (first voice engine then
Chromium). It's more logical for this burden to fall on AudioProcessing,
however, who can be expected to know that this is a reasonable approach
for AEC. Permitting two render channels results in running two AECs
serially.
Critically, in my recent change to have Chromium adopt the float
interface:
https://codereview.chromium.org/420603004
I removed the downmixing by Chromium, forgetting that we hadn't yet
enabled this feature in AudioProcessing. This corrects that oversight.
The change in paths hit by production users is very minor. As commented
it required adding downmixing to the int16_t path to satisfy
bit-exactness tests.
For reference, find the ApmTest.Process errors here:
https://paste.googleplex.com/6372007910309888
BUG=webrtc:3853
TESTED=listened to the files output from the Process test, and verified
that they sound as expected: higher echo while the AEC is adapting, but
afterwards very close.
R=aluebs@webrtc.org, bjornv@webrtc.org, kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/31459004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7292 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/audio_buffer.cc
|
| e364ac902fe7e711de73e2fde22dff3cbe1e3a6f |
|
18-Jul-2014 |
kwiberg@webrtc.org <kwiberg@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
AudioBuffer: Optimize const accesses to arrays that autoconvert int16<->float
Specifically, when someone asks for a const pointer to the int16
version of the array, there's no need to invalidate the float version
of that array, and vice versa. (But obviously, invalidation still has
to happen when someone asks for a non-const pointer.)
R=aluebs@webrtc.org, andrew@webrtc.org, minyue@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/18809004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6725 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/audio_buffer.cc
|
| 2b6bc8d84f309c27fb5264145fb2a5c87cb2267a |
|
17-Jul-2014 |
kwiberg@webrtc.org <kwiberg@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
AudioBuffer: Eliminate the SplitChannelBuffer class
It's just a container for two IFChannelBuffers, and doesn't earn its
keep. The main problem is that the number of methods it needs that
just forward calls to either of its two IFChannelBuffers was already
large, and was about to grow.
R=aluebs@webrtc.org, minyue@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16919004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6717 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/audio_buffer.cc
|
| 2561d524600b6bcfd0c4a24621cebdcdf6d027f8 |
|
17-Jul-2014 |
aluebs@webrtc.org <aluebs@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Simplify AudioBuffer::mixed_low_pass_data API
R=andrew@webrtc.org, kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/21869004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6715 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/audio_buffer.cc
|
| af93fc08a19e75bc0b3fab393345c29b2094391d |
|
17-Jul-2014 |
kwiberg@webrtc.org <kwiberg@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
AudioBuffer: Let ChannelBuffer handle bounds checking of channel parameter
R=aluebs@webrtc.org, minyue@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13019004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6714 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/audio_buffer.cc
|
| 38214d53db5f19228b38424700db8b90cf8b0234 |
|
03-Jul-2014 |
kwiberg@webrtc.org <kwiberg@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
EchoCancellationImpl::ProcessRenderAudio: Use float samples directly
This patch lets EchoCancellationImpl::ProcessRenderAudio ask the given
AudioBuffer for float sample data directly, instead of asking for
int16 samples and then converting manually.
Since EchoCancellationImpl::ProcessRenderAudio takes a const
AudioBuffer*, it was necessary to add some const accessors for float
data to AudioBuffer.
R=aluebs@webrtc.org, andrew@webrtc.org, bjornv@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14749004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6590 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/audio_buffer.cc
|
| 8e4401b5a0cdbb33d7d7fb2547464724411c3ee2 |
|
03-Jun-2014 |
kwiberg@webrtc.org <kwiberg@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Reformat integer accessors to look like their float counterparts
The new format is at least as easy to read, and takes less space.
BUG=
R=aluebs@webrtc.org, bjornv@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16539004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6311 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/audio_buffer.cc
|
| c0035a67a15943219f9899b703332feb9ffdba90 |
|
03-Jun-2014 |
kwiberg@webrtc.org <kwiberg@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Remove an optimization that's no longer worth the extra complexity it causes
The data_ optimization was a way to operate on the data directly
instead of copying it, applicable in the mono, non-float case. Since a
few audio_processing steps are already using floats (with more
hopefully to come), we don't end up benefiting from the optimization
anyway, so we might as well remove it.
BUG=
R=aluebs@webrtc.org, bjornv@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/15539004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6307 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/audio_buffer.cc
|
| d5da25063c46678f50882da5c1bad21ca3cfe271 |
|
15-May-2014 |
mflodman@webrtc.org <mflodman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert "Revert "Audio processing: Feed each processing step its choice
of int or float data"
This reverts commit 6142.
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17519004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6172 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/audio_buffer.cc
|
| 21299d4e00781e199a53ba33ec192cdce920acec |
|
14-May-2014 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Remove the use of AudioFrame::energy_ from AudioProcessing and VoE.
We want to remove energy_ entirely as we've seen that carrying around
this potentially invalid value is dangerous.
Results in the removal of AudioBuffer::is_muted(). This wasn't used in
practice any longer, after the level calculation moved directly to
channel.cc
Instead, now use ProcessMuted() in channel.cc, to shortcut the level
computation when the signal is muted.
BUG=3315
TESTED=Muting the channel in voe_cmd_test results in rms=127.
R=bjornv@webrtc.org, kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12529004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6159 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/audio_buffer.cc
|
| b1a66d166c137aef483b1a24540b00b0d45bdf27 |
|
14-May-2014 |
mflodman@webrtc.org <mflodman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert "Audio processing: Feed each processing step its choice of int or float data"
This reverts r6138.
tbr=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13509004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6142 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/audio_buffer.cc
|
| 934a265a47f71f06fea5878bd50ddcc629ca455a |
|
14-May-2014 |
kwiberg@webrtc.org <kwiberg@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Audio processing: Feed each processing step its choice of int or float data
Each audio processing step is given a pointer to an AudioBuffer, where
it can read and write int data. This patch adds corresponding
AudioBuffer methods to read and write float data; the buffer will
automatically convert the stored data between int and float as
necessary.
This patch also modifies the echo cancellation step to make use of the
new methods (it was already using floats internally; now it doesn't
have to convert from and to ints anymore).
(The reference data to the ApmTest.Process test had to be modified
slightly; this is because the echo canceller no longer unnecessarily
converts float data to int and then immediately back to float for each
iteration in the loop in EchoCancellationImpl::ProcessCaptureAudio.)
BUG=
R=aluebs@webrtc.org, andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/18399005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6138 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/audio_buffer.cc
|
| 4cc763621eeeb29d0bf1d16d69b2f96d711ead2b |
|
08-May-2014 |
kwiberg@webrtc.org <kwiberg@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
AudioBuffer: Eliminate data_was_mixed_, and document what's left of data_
data_was_mixed_ was always false, so it can be removed. That makes the
role of data_ simpler, but not so simple that it doesn't merit an
explanation.
BUG=
R=aluebs@webrtc.org, andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17409004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6076 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/audio_buffer.cc
|
| 65f933899b815b6c09f8ca7beefeace09ee3ae70 |
|
30-Apr-2014 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fix constness of AudioBuffer accessors.
Don't return non-const pointers from const accessors and deal with the
spillover. Provide overloaded versions as needed.
Inspired by kwiberg:
https://webrtc-codereview.appspot.com/12379005/
R=bjornv@webrtc.org, kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/15379004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6030 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/audio_buffer.cc
|
| 103657b48442dedd1742fca4a73d5131bf4ae624 |
|
24-Apr-2014 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add keyboard channel support to AudioBuffer.
Also use local aliases for AudioBuffers for brevity.
BUG=2894
R=aluebs@webrtc.org, bjornv@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13369005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5973 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/audio_buffer.cc
|
| ddbb8a2c243f9d54cb0ce0092e341dfc6e126bb3 |
|
22-Apr-2014 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Support arbitrary input/output rates and downmixing in AudioProcessing.
Select "processing" rates based on the input and output sampling rates.
Resample the input streams to those rates, and if necessary to the
output rate.
- Remove deprecated stream format APIs.
- Remove deprecated device sample rate APIs.
- Add a ChannelBuffer class to help manage deinterleaved channels.
- Clean up the splitting filter state.
- Add a unit test which verifies the output against known-working
native format output.
BUG=2894
R=aluebs@webrtc.org, bjornv@webrtc.org, xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/9919004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5959 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/audio_buffer.cc
|
| d2f366f28ceb738a9f193a5ce1868b50b3697197 |
|
17-Apr-2014 |
kwiberg@webrtc.org <kwiberg@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
StereoToMono: Remove useless call to WebRtcSpl_SatW32ToW16
The max value is ((2**15 - 1) + (2**15 - 1)) >> 1
== (2**16 - 2) >> 1
== 2**15 - 1
which doesn't overflow.
The min value is (-2**15 + -2**15) >> 1
== -2**16 >> 1
== -2**15
which doesn't overflow.
Since those two bracket all possible results, the call to
WebRtcSpl_SatW32ToW16 is redundant.
BUG=
R=andrew@webrtc.org, bjornv@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12019004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5929 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/audio_buffer.cc
|
| 17e40641b30559602e26382e500bd9708bad37e3 |
|
04-Mar-2014 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add a deinterleaved float interface to AudioProcessing.
This is mainly to support the native audio format in Chrome. Although
this implementation just moves the float->int conversion under the hood,
we will transition AudioProcessing towards supporting this format
throughout.
- Add a test which verifies we get identical output with the float and
int interfaces.
- The float and int wrappers are tasked with conversion to the
AudioBuffer format. A new shared Process/Analyze method does most of
the work.
- Add a new field to the debug.proto to hold deinterleaved data.
- Add helpers to audio_utils.cc, and start using numeric_limits.
- Note that there was no performance difference between numeric_limits
and a literal value when measured on Linux using gcc or clang.
BUG=2894
R=aluebs@webrtc.org, bjornv@webrtc.org, henrikg@webrtc.org, tommi@webrtc.org, turaj@webrtc.org, xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/9179004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5641 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/audio_buffer.cc
|
| 7fad4b8c9f1e9a6e3de9962fb74d4953b4f1bb03 |
|
28-May-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Include files from webrtc/.. paths in audio_processing/
BUG=1662
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1559004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4116 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/audio_buffer.cc
|
| b7192b82476d00384fdc153e6a09a6ac53cef67b |
|
10-Apr-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
WebRtc_Word32 -> int32_t in audio_processing/
BUG=314
Review URL: https://webrtc-codereview.appspot.com/1307004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3809 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/audio_buffer.cc
|
| 14b43beb7ce4440b30dcea31196de5b4a529cb6b |
|
22-Oct-2012 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Move src/ -> webrtc/
TBR=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/915006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@2963 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_processing/audio_buffer.cc
|